Notes from 'CCNA Voice 640-461 Official Certification Guide - 2nd edition by Jeremy Cioara and Michael Valentine

Table of Contents

Trad Voice vs Unified Voice

Analog

Supervisory signaling - on hook, off hook, ringing
information signaling - dial tone, busy, ringback, ...
address signaling - dtmf, pulse

Digital

TDM
T1 from is 193 bits
CAS (Channel associated signaling - steal bits from voice bw)
    T1s use 8th bit on every 5th sample in each DS0
CCS (Common channel signaling - dedicated signaling channel)
    24th slot is used on T1s
    17th slot used on E1s

e164

Max of 15 digits in length
country code, national dest code, subscriber number
NANP breaks down to
- country code
- area code
- CO/Exchange code
- station code

VoIP

Benefits

- reduced cost of communicating
- reduced cost of cabling
- seamless voice networks
- take phone with you
- IP softphones
- unified email voicemail fax
- increased productivity - multi device ring
- feature rich comm
- open compatible standards

Voice packet conversion

avg human ear hears frequencies form 20-20,000 Hz
human speech uses freqs from 200-9000 Hz
telephone channels typically transmit freqs from 300-3400
Nyquist recreating 300-4000Hz - sample at 2x the highest frequency
  8000 samples per seconds x 8 bits each 64kbs
alaw and ulaw are exact opposites 0s and 1s
MOS 1-5 'nowadays, a chicken leg is a rare dish'
Codec bw       MOS typical
G711  64kbps   4.1
ILBC  15.2kbps 4.1
G729  8kbps    3.92
G726  32kbps   3.85
G729a 9kbps    3.7
G728  16kbps   3.61
 G729a - sacrifices audio qual to achive more proc efficient coding
 G729b - supports VAD
  MOS can be bad...

DSPs

calc - http://www.cisco.com/web/applicat/dsprecal/index.html 
PVDM2-8  - .5 DSP chip
PVDM2-16 - 1 DSP chip
PVDM2-32 - 2 DSP chips
PVDM2-48 - 3 DSP chips
PVDM2-64 - 4 DSP chips

DSP resources can handle double # of med-complex calls per dsp as high
Med Complex - G711 G726 G729a, G729ab
Hi Complex  - G728 G723 G729 G729b iLBC

RTP/RTCP

RTP even # UDP port
header - payload type, seq number, time stamp
UDP port 16384-32767 by default
1 way (need 2 for conversation)
RTCP reports stats odd number port above RTP
- packet count, delay, loss, and jitter
- gets sent at least 1x per every 5 seconeds
- CME can log and report info

Cisco UC

CME

CUE

CUCM

CUCM cluster communication

CUCM Business Edition

Up to 500 IP phones no server redundancy (clustering)

Unity Connection

CUP

Unified Presence

CUPC

Cisco Unified Personal Communicator - IM client with soft phone, presence, IM, visual voicemail, employee directory, comm history, video, web conferencing. LDAP auth, change, multiuser chat, peristent chat (in 'rooms'). Can see on/off hock on phones.

Phone concepts

Plugs in back

POE

VLANs

Broadcast domain, IP Subnet 
config term
vlan 10
    name VOICE
vlan 50
    name DATA

int range fa0/2 - 24
    switchport mode access
    spanning-tree portfast
    switchport access vlan 50
    switchport voice vlan 10
With this config, only an attached Cisco IP Phone will be able to access the voice VLAN. Learns from it via CDP. non-Cisco IP phones have to be manually configured with voice VLAN number

Cisco IP phone boot proc

  1. PoE (Cisco proprietary or 802.3af
  2. switch delivers voice vlan info via CDP
  3. DHCP req asking for IP address on voice vlan
  4. DHCP response including addr, domain, gw, option 150 tftp svr
  5. contacts TFTP server and downloads cfg file, including list of valid call processing agents
  6. phone attempts to contact processing svr (primary svr) listed in its cfg file. If 1st fails, go to 2nd, than 3rd.
  7. IP phone ids itself by its mac addr. Call proc svr matches in db, and sends operating cfg to phone (dir/line number, ring tone, softkey layout, etc). Sent using SIP or SCCP?

Configuring Rtr based DHCP svc

config term
ip dhcp excluded-address 10.1.1.1 10.1.1.9
ip dhcp excluded-address 10.1.2.1 10.1.2.9
ip dhcp pool DATASCOPE
    network 10.1.2.0 255.255.255.0
    default-router 10.1.2.1
    dns-server 10.1.3.10
ip dhcp pool VOICESCOPE
    network 10.1.1.0 255.255.255.0
    default-router 10.1.1.1
    option 150 ip 10.1.3.20 (if using CME, make this
                                     same as default-router?)
    dns-server 10.1.3.10
    
'if you have a central dhcp svr use
ip helper-address <DHCP server IP address>

Configuring Rtr to receive time via NTP

config term
ntp server 10.1.3.30
clock timezone NYC -5 (assuming source is UTC)
ntp master 4 (deliver data/time info to requesting clients
                                    marking it with stratum 
                                    number 4)
                clients could be phones or CME
exit
show ntp associations

CME

CCP, CME GUI, and CLI overview

Endpoints and End users - CME

tftp svr cfg

Point CME TO Ext TFTP Svr

cnf-file location tftp:// < ip address of TFTP server >
(from telephony service configuration mode.

User CME as TFTP

dir flash:/phone/7941
tftp-server flash:/phone/7941/<loadname>.loads alias <loadname>.loads (redirects load tftp reqs to path
	make sure all files available via 'alias' command

Base CME Cfg

telephony-service
ip source-addr <ipaddr>
max-ephones 15
max-dn 30

ephone-dn 1 (from 1-150, but lower than max-dn)
    number 1200
ephone-dn 2 dual-line
    number 1201
ephone-dn 3 dual-line
    number 1202 secondary 6175551202

ephone 1
    mac-addr 0123.4567.8901
    button 1:2 (button 1 uses ephone-dn 2)
               (: indicates normal ring)
    button 2:3 3:1
    restart    (restart phone (reboot) and download cfg from tftp)

show ephone
<...list of ephone params...>

User Cfgs in CME

ephone 1
    username Joe password Dapassword
    pin 555121
    button 1:1
    restart
    exit 
ephone-dn 1
    name Sarah Smith
    exit

CME CCP Config (GUI)

Cisco Config Professional
Go to Configure > Unified Communications > Telephony Settings to configure max-ephones, max-dn, ip source addr, etc.
Click 'Refresh' button to force sync of config on box.
CCP will show command line config to match up with your config
Bulk import wizard to the right (allows import of CSV files)
Config Extensions
Configure > Unified Communications > Users, Phones and Extensions > Phones
User Settings/Names
Creates user db which can be searched on phone as local directory
Configure > Unified Communications > Users, Phones and Extensions > User Settings
Click Phones/Extensions tab at top of the Create User window to associate phone, extensions and user account.
Type free form IOS commands
Configure > View > IOS Show Commands

CME Dialplan

show voice port summary (ephone-dns show up as EXFS ports)
3 common areas of config

FXS

voice-port 0/0/0
  signal loopStart (or groundStart) - SIGNALING
  cptone US (or some other 2 letter country code) - CALL PROGRESS TONES
  station-id name Joe Smith - CALLER ID INFO
  station-id number 8005551234 - CALLER ID INFO

FXO

voice-port 0/1/0
  signal loopStart (or groundStart) - SIGNALING
  station-id name Joe Smith - CALLER ID INFO
  station-id number 8005551234 - CALLER ID INFO
  dial-type dtmf (or pulse)
  ring number (number of rings before pick up)

Digital circuits

show controllers t1

controller t1 1/0
  framing esf (or sf)
  linecode b8zs (or ami)
  clock source line (or free-running or internal (provide clocking))
  ds0-group 1 timeslots 1-12 type e&m-wink-start
  ds0-group 2 timeslots 13-24 type fxo-loop-start
ds0-group commands automatically creates voice ports PSTN carrier typically uses FXO loop start signaling
PBXes often support one of the E&M signaling types

show voice port summary will show port:ds0group (e.g. 1/0:2 for channels 13-24 as fxo-loop-start

isdn switch-type primary-dms100 (or primary-5ess or ...)
controller t1 1/1
  pri-group timeslots 1-24
pri-group commands automatically creates voice ports

show voice port summary will list channels 01-23 associated with port 1/0:23
  These commands will not show up in 'show ip interface brief' 
channel 23 (time slot 24) is signaling on T1 ISDN
channel 16 (time slot 16) is signaling on E1 ISDN
make sure you have enough DSPs to support

Dial Plan

POTS Dial peers

dial-peer voice 1101 pots
  destination-pattern 1101
  port 0/0/0 (probably a FXS)

dial-peer voice 2000 pots
  destination-pattern 2...
  no digit-strip (do not strip any (POTS) explicitly defined digits (e.g. the 2 above)
  port 1/0:23 (pri)

dial-peer voice 10 pots
  destination-pattern [2-9]..[2-9]......
  prefix 1 (prefix 1 before outgoing dest pattern)
  port 1/0:1

dial-peer voice 9011 pots
  destination-pattern 91[2-9]..[2-9]......
  forward-digits 11 (right justified will forward from 1 onwards)
  port 1/0:1

num-exp 4... 5... (replace all 4 digit numbers starting with 4 to
                          4 digit numbers starting wiht 5

to change type of dial peer (e.g. to voip) 'no dial-peer voice 1101' or whatever tag...
show dial-peer voice (for details on each dial-peer)
show dial-peer voice summary
           AD                              PRE PASS                OUT
TAG  TYPE  MIN OPER  PREFIX  DEST-PATTERN  FER THRU  SESS-TARGET   STAT PORT
1101 pots  up  up            1101          0                     up   0/0/0

debug voip dialpeer (to analyze digit processing)

show dialplan number <number>
  (shows all matching dial peers in order in which router will use them...
   more specific matches are 1st)

VoIP Dial peers

dial-peer voice 2001 voip
  destination pattern 3...
  session target ipv4:<ipaddr> (or dns:<name>)
  codec g711ulaw (or g711alaw, g729r8, etc - if codec doesn't match - reorder)

dial peer wildcards

.matches any dial digits 0-9 or *
+matches one or more instances of preceding digit
[]matches range of digits. ^in here designates 'do not match'
Tmatches any number of dialed digits from 0-32 digits
,inserts a 1-second pause between dialed digits
# is not a wild card but forces immediate processing of dialed number
if you use T, recommended to use .T. which requies user to dial at least one digit. Otherwise no digits dial could match on T302 interdigit timeout

NA sample PSTN dest patterns

[2-9]..[2-9]......10 digit dialing
1[2-9]..[2-9]......1+10 digit dialing areas
[469]11service number 411, 611, 911
011Tinternational dialing

PLARs

voice-port 0/0/0
  connection plar 1102 (when someone picks up phone - dial 1102)

inbound dial peers

match inbound dial peers through following 5 methods
  1. incoming called-number - DNIS
  2. answer-address - ANI
  3. destination-pattern - ANI
  4. port dial-peer - incoming POTS dial peer
  5. use dial peer 0

dial peer 0